From 0cb612b43bc1ef42cde8cb3887dc48917d5a58dd Mon Sep 17 00:00:00 2001 From: "jmarusic@webrtc.org" Date: Tue, 17 Mar 2015 12:12:17 +0000 Subject: [PATCH] We changed Encode() and EncodeInternal() return type from bool to void in this issue: https://webrtc-codereview.appspot.com/38279004/ Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43839004 Cr-Commit-Position: refs/heads/master@{#8749} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../audio_coding/codecs/audio_encoder.cc | 19 ++++--- .../audio_coding/codecs/audio_encoder.h | 26 ++++----- .../codecs/cng/audio_encoder_cng.cc | 54 +++++++++--------- .../codecs/cng/audio_encoder_cng_unittest.cc | 51 +++++++---------- .../codecs/cng/include/audio_encoder_cng.h | 17 ++---- .../codecs/g711/audio_encoder_pcm.cc | 21 +++---- .../codecs/g711/include/audio_encoder_pcm.h | 9 ++- .../codecs/g722/audio_encoder_g722.cc | 21 +++---- .../codecs/g722/include/audio_encoder_g722.h | 9 ++- .../codecs/ilbc/audio_encoder_ilbc.cc | 23 ++++---- .../ilbc/interface/audio_encoder_ilbc.h | 9 ++- .../codecs/isac/audio_encoder_isac_t.h | 9 ++- .../codecs/isac/audio_encoder_isac_t_impl.h | 20 ++++--- .../codecs/mock/mock_audio_encoder.h | 11 ++-- .../codecs/opus/audio_encoder_opus.cc | 26 ++++----- .../opus/interface/audio_encoder_opus.h | 9 ++- .../codecs/red/audio_encoder_copy_red.cc | 56 +++++++++---------- .../codecs/red/audio_encoder_copy_red.h | 9 ++- .../red/audio_encoder_copy_red_unittest.cc | 50 ++++++++--------- .../main/acm2/acm_generic_codec.cc | 4 +- .../neteq/audio_decoder_unittest.cc | 6 +- 21 files changed, 222 insertions(+), 237 deletions(-) diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc index 1d83e54a0..76cb33b69 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc @@ -19,16 +19,19 @@ AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() { AudioEncoder::EncodedInfo::~EncodedInfo() { } -void AudioEncoder::Encode(uint32_t rtp_timestamp, - const int16_t* audio, - size_t num_samples_per_channel, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +const AudioEncoder::EncodedInfo AudioEncoder::kZeroEncodedBytes; + +AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp, + const int16_t* audio, + size_t num_samples_per_channel, + size_t max_encoded_bytes, + uint8_t* encoded) { CHECK_EQ(num_samples_per_channel, static_cast(SampleRateHz() / 100)); - EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info); - CHECK_LE(info->encoded_bytes, max_encoded_bytes); + EncodedInfo info = + EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded); + CHECK_LE(info.encoded_bytes, max_encoded_bytes); + return info; } int AudioEncoder::RtpTimestampRateHz() const { diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h index 20ac8b922..6e29f08c3 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h @@ -54,20 +54,21 @@ class AudioEncoder { std::vector redundant; }; + static const EncodedInfo kZeroEncodedBytes; + virtual ~AudioEncoder() {} // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 * // num_channels() samples). Multi-channel audio must be sample-interleaved. - // The encoder produces zero or more bytes of output in |encoded|, - // and provides additional encoding information in |info|. + // The encoder produces zero or more bytes of output in |encoded| and + // returns additional encoding information. // The caller is responsible for making sure that |max_encoded_bytes| is // not smaller than the number of bytes actually produced by the encoder. - void Encode(uint32_t rtp_timestamp, - const int16_t* audio, - size_t num_samples_per_channel, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info); + EncodedInfo Encode(uint32_t rtp_timestamp, + const int16_t* audio, + size_t num_samples_per_channel, + size_t max_encoded_bytes, + uint8_t* encoded); // Return the input sample rate in Hz and the number of input channels. // These are constants set at instantiation time. @@ -107,11 +108,10 @@ class AudioEncoder { virtual void SetProjectedPacketLossRate(double fraction) {} protected: - virtual void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) = 0; + virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) = 0; }; } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc index d7c1ea013..38ebca13e 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc @@ -109,13 +109,12 @@ void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) { speech_encoder_->SetProjectedPacketLossRate(fraction); } -void AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { CHECK_GE(max_encoded_bytes, static_cast(num_cng_coefficients_ + 1)); - info->encoded_bytes = 0; const int num_samples = SampleRateHz() / 100 * NumChannels(); if (speech_buffer_.empty()) { CHECK_EQ(frames_in_buffer_, 0); @@ -126,7 +125,7 @@ void AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, } ++frames_in_buffer_; if (frames_in_buffer_ < speech_encoder_->Num10MsFramesInNextPacket()) { - return; + return kZeroEncodedBytes; } CHECK_LE(frames_in_buffer_ * 10, kMaxFrameSizeMs) << "Frame size cannot be larger than " << kMaxFrameSizeMs @@ -159,14 +158,15 @@ void AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz()); } + EncodedInfo info; switch (activity) { case Vad::kPassive: { - EncodePassive(max_encoded_bytes, encoded, info); + info = EncodePassive(max_encoded_bytes, encoded); last_frame_active_ = false; break; } case Vad::kActive: { - EncodeActive(max_encoded_bytes, encoded, info); + info = EncodeActive(max_encoded_bytes, encoded); last_frame_active_ = true; break; } @@ -178,15 +178,17 @@ void AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp, speech_buffer_.clear(); frames_in_buffer_ = 0; + return info; } -void AudioEncoderCng::EncodePassive(size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive( + size_t max_encoded_bytes, + uint8_t* encoded) { bool force_sid = last_frame_active_; bool output_produced = false; const size_t samples_per_10ms_frame = SamplesPer10msFrame(); CHECK_GE(max_encoded_bytes, frames_in_buffer_ * samples_per_10ms_frame); + AudioEncoder::EncodedInfo info; for (int i = 0; i < frames_in_buffer_; ++i) { int16_t encoded_bytes_tmp = 0; CHECK_GE(WebRtcCng_Encode(cng_inst_.get(), @@ -195,30 +197,32 @@ void AudioEncoderCng::EncodePassive(size_t max_encoded_bytes, encoded, &encoded_bytes_tmp, force_sid), 0); if (encoded_bytes_tmp > 0) { CHECK(!output_produced); - info->encoded_bytes = static_cast(encoded_bytes_tmp); + info.encoded_bytes = static_cast(encoded_bytes_tmp); output_produced = true; force_sid = false; } } - info->encoded_timestamp = first_timestamp_in_buffer_; - info->payload_type = cng_payload_type_; - info->send_even_if_empty = true; - info->speech = false; + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = cng_payload_type_; + info.send_even_if_empty = true; + info.speech = false; + return info; } -void AudioEncoderCng::EncodeActive(size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive( + size_t max_encoded_bytes, + uint8_t* encoded) { const size_t samples_per_10ms_frame = SamplesPer10msFrame(); + AudioEncoder::EncodedInfo info; for (int i = 0; i < frames_in_buffer_; ++i) { - speech_encoder_->Encode(first_timestamp_in_buffer_, - &speech_buffer_[i * samples_per_10ms_frame], - samples_per_10ms_frame, max_encoded_bytes, - encoded, info); + info = speech_encoder_->Encode( + first_timestamp_in_buffer_, &speech_buffer_[i * samples_per_10ms_frame], + samples_per_10ms_frame, max_encoded_bytes, encoded); if (i < frames_in_buffer_ - 1) { - CHECK_EQ(info->encoded_bytes, 0u) << "Encoder delivered data too early."; + CHECK_EQ(info.encoded_bytes, 0u) << "Encoder delivered data too early."; } } + return info; } size_t AudioEncoderCng::SamplesPer10msFrame() const { diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc index 5dfa4d545..528cf343e 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc @@ -75,9 +75,8 @@ class AudioEncoderCngTest : public ::testing::Test { void Encode() { ASSERT_TRUE(cng_) << "Must call CreateCng() first."; - encoded_info_ = AudioEncoder::EncodedInfo(); - cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_, - encoded_.size(), &encoded_[0], &encoded_info_); + encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_, + encoded_.size(), &encoded_[0]); timestamp_ += num_audio_samples_10ms_; } @@ -92,24 +91,24 @@ class AudioEncoderCngTest : public ::testing::Test { .WillRepeatedly(Return(active_speech ? Vad::kActive : Vad::kPassive)); // Don't expect any calls to the encoder yet. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0); for (int i = 0; i < blocks_per_frame - 1; ++i) { Encode(); EXPECT_EQ(0u, encoded_info_.encoded_bytes); } - AudioEncoder::EncodedInfo info; if (active_speech) { // Now expect |blocks_per_frame| calls to the encoder in sequence. // Let the speech codec mock return true and set the number of encoded // bytes to |kMockReturnEncodedBytes|. InSequence s; for (int j = 0; j < blocks_per_frame - 1; ++j) { - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(AudioEncoder::kZeroEncodedBytes)); } + AudioEncoder::EncodedInfo info; info.encoded_bytes = kMockReturnEncodedBytes; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(info)); } Encode(); if (active_speech) { @@ -254,7 +253,7 @@ TEST_F(AudioEncoderCngTest, EncodePassive) { EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) .WillRepeatedly(Return(Vad::kPassive)); // Expect no calls at all to the speech encoder mock. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0); uint32_t expected_timestamp = timestamp_; for (int i = 0; i < 100; ++i) { Encode(); @@ -284,20 +283,23 @@ TEST_F(AudioEncoderCngTest, MixedActivePassive) { CreateCng(); // All of the frame is active speech. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .Times(6); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .Times(6) + .WillRepeatedly(Return(AudioEncoder::kZeroEncodedBytes)); EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kActive)); EXPECT_TRUE(encoded_info_.speech); // First half of the frame is active speech. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .Times(6); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .Times(6) + .WillRepeatedly(Return(AudioEncoder::kZeroEncodedBytes)); EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kPassive)); EXPECT_TRUE(encoded_info_.speech); // Second half of the frame is active speech. - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .Times(6); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .Times(6) + .WillRepeatedly(Return(AudioEncoder::kZeroEncodedBytes)); EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive)); EXPECT_TRUE(encoded_info_.speech); @@ -336,22 +338,10 @@ TEST_F(AudioEncoderCngTest, VadInputSize60Ms) { CheckVadInputSize(60, 30, 30); } -// Verifies that the EncodedInfo struct pointer passed to -// AudioEncoderCng::Encode is propagated to the Encode call to the underlying -// speech encoder. -TEST_F(AudioEncoderCngTest, VerifyEncoderInfoPropagation) { - CreateCng(); - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, &encoded_info_)); - EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1)); - EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) - .WillOnce(Return(Vad::kActive)); - Encode(); -} - // Verifies that the correct payload type is set when CNG is encoded. TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) { CreateCng(); - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)).Times(0); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0); EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1)); EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)) .WillOnce(Return(Vad::kPassive)); @@ -385,8 +375,7 @@ TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) { .WillOnce(Return(Vad::kActive)); AudioEncoder::EncodedInfo info; info.encoded_bytes = kMockReturnEncodedBytes; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).WillOnce(Return(info)); Encode(); EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes); diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h index cc0165054..daecd51ff 100644 --- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h +++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h @@ -56,11 +56,10 @@ class AudioEncoderCng final : public AudioEncoder { void SetProjectedPacketLossRate(double fraction) override; protected: - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; private: // Deleter for use with scoped_ptr. E.g., use as @@ -69,12 +68,8 @@ class AudioEncoderCng final : public AudioEncoder { inline void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); } }; - void EncodePassive(size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info); - void EncodeActive(size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info); + EncodedInfo EncodePassive(size_t max_encoded_bytes, uint8_t* encoded); + EncodedInfo EncodeActive(size_t max_encoded_bytes, uint8_t* encoded); size_t SamplesPer10msFrame() const; AudioEncoder* speech_encoder_; diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc index 5c43a85cb..99566aacb 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -66,11 +66,11 @@ int AudioEncoderPcm::Max10MsFramesInAPacket() const { return num_10ms_frames_per_packet_; } -void AudioEncoderPcm::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { const int num_samples = SampleRateHz() / 100 * NumChannels(); if (speech_buffer_.empty()) { first_timestamp_in_buffer_ = rtp_timestamp; @@ -79,17 +79,18 @@ void AudioEncoderPcm::EncodeInternal(uint32_t rtp_timestamp, speech_buffer_.push_back(audio[i]); } if (speech_buffer_.size() < full_frame_samples_) { - info->encoded_bytes = 0; - return; + return kZeroEncodedBytes; } CHECK_EQ(speech_buffer_.size(), full_frame_samples_); CHECK_GE(max_encoded_bytes, full_frame_samples_); int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded); CHECK_GE(ret, 0); speech_buffer_.clear(); - info->encoded_timestamp = first_timestamp_in_buffer_; - info->payload_type = payload_type_; - info->encoded_bytes = static_cast(ret); + EncodedInfo info; + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = payload_type_; + info.encoded_bytes = static_cast(ret); + return info; } int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h index e64bcea27..6e588ecfc 100644 --- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h +++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h @@ -41,11 +41,10 @@ class AudioEncoderPcm : public AudioEncoder { protected: AudioEncoderPcm(const Config& config, int sample_rate_hz); - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; virtual int16_t EncodeCall(const int16_t* audio, size_t input_len, diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc index bbdcb1da7..7eb44933a 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc @@ -77,11 +77,11 @@ int AudioEncoderG722::Max10MsFramesInAPacket() const { return num_10ms_frames_per_packet_; } -void AudioEncoderG722::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { CHECK_GE(max_encoded_bytes, MaxEncodedBytes()); if (num_10ms_frames_buffered_ == 0) @@ -95,8 +95,7 @@ void AudioEncoderG722::EncodeInternal(uint32_t rtp_timestamp, // If we don't yet have enough samples for a packet, we're done for now. if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { - info->encoded_bytes = 0; - return; + return kZeroEncodedBytes; } // Encode each channel separately. @@ -124,9 +123,11 @@ void AudioEncoderG722::EncodeInternal(uint32_t rtp_timestamp, encoded[i * num_channels_ + j] = interleave_buffer_[2 * j] << 4 | interleave_buffer_[2 * j + 1]; } - info->encoded_bytes = samples_per_channel / 2 * num_channels_; - info->encoded_timestamp = first_timestamp_in_buffer_; - info->payload_type = payload_type_; + EncodedInfo info; + info.encoded_bytes = samples_per_channel / 2 * num_channels_; + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = payload_type_; + return info; } int AudioEncoderG722::SamplesPerChannel() const { diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h index 81b44d625..b1be6b952 100644 --- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h +++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h @@ -38,11 +38,10 @@ class AudioEncoderG722 : public AudioEncoder { int Max10MsFramesInAPacket() const override; protected: - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; private: // The encoder state for one channel. diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc index 1e85a077f..4971e7ba9 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -63,11 +63,11 @@ int AudioEncoderIlbc::Max10MsFramesInAPacket() const { return num_10ms_frames_per_packet_; } -void AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes()); // Save timestamp if starting a new packet. @@ -82,8 +82,7 @@ void AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp, // If we don't yet have enough buffered input for a whole packet, we're done // for now. if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { - info->encoded_bytes = 0; - return; + return kZeroEncodedBytes; } // Encode buffered input. @@ -95,10 +94,12 @@ void AudioEncoderIlbc::EncodeInternal(uint32_t rtp_timestamp, kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded); CHECK_GE(output_len, 0); - info->encoded_bytes = output_len; - DCHECK_EQ(info->encoded_bytes, RequiredOutputSizeBytes()); - info->encoded_timestamp = first_timestamp_in_buffer_; - info->payload_type = payload_type_; + EncodedInfo info; + info.encoded_bytes = output_len; + DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes()); + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = payload_type_; + return info; } size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const { diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h index a5378d1ed..91d17b4f8 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h +++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h @@ -38,11 +38,10 @@ class AudioEncoderIlbc : public AudioEncoder { int Max10MsFramesInAPacket() const override; protected: - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; private: size_t RequiredOutputSizeBytes() const; diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index 6b197bc2d..95c83e808 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -84,11 +84,10 @@ class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder { protected: // AudioEncoder protected method. - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; // AudioDecoder protected method. int DecodeInternal(const uint8_t* encoded, diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h index 87d71ab38..02acfa6dd 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h @@ -184,11 +184,11 @@ int AudioEncoderDecoderIsacT::Max10MsFramesInAPacket() const { } template -void AudioEncoderDecoderIsacT::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { CriticalSectionScoped cs_lock(lock_.get()); if (!packet_in_progress_) { // Starting a new packet; remember the timestamp for later. @@ -206,15 +206,17 @@ void AudioEncoderDecoderIsacT::EncodeInternal(uint32_t rtp_timestamp, // buffer. All we can do is check for an overrun after the fact. CHECK(static_cast(r) <= max_encoded_bytes); - info->encoded_bytes = r; if (r == 0) - return; + return kZeroEncodedBytes; // Got enough input to produce a packet. Return the saved timestamp from // the first chunk of input that went into the packet. packet_in_progress_ = false; - info->encoded_timestamp = packet_timestamp_; - info->payload_type = payload_type_; + EncodedInfo info; + info.encoded_bytes = r; + info.encoded_timestamp = packet_timestamp_; + info.payload_type = payload_type_; + return info; } template diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h index 7425e9a18..25fd7a837 100644 --- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h @@ -29,12 +29,11 @@ class MockAudioEncoder : public AudioEncoder { MOCK_METHOD1(SetTargetBitrate, void(int)); MOCK_METHOD1(SetProjectedPacketLossRate, void(double)); // Note, we explicitly chose not to create a mock for the Encode method. - MOCK_METHOD5(EncodeInternal, - void(uint32_t timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info)); + MOCK_METHOD4(EncodeInternal, + EncodedInfo(uint32_t timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded)); }; } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index ae08423db..be92589ce 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -183,19 +183,18 @@ void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) { } } -void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { +AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { if (input_buffer_.empty()) first_timestamp_in_buffer_ = rtp_timestamp; input_buffer_.insert(input_buffer_.end(), audio, audio + samples_per_10ms_frame_); if (input_buffer_.size() < (static_cast(num_10ms_frames_per_packet_) * samples_per_10ms_frame_)) { - info->encoded_bytes = 0; - return; + return kZeroEncodedBytes; } CHECK_EQ(input_buffer_.size(), static_cast(num_10ms_frames_per_packet_) * @@ -207,12 +206,13 @@ void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp, ClampInt16(max_encoded_bytes), encoded); CHECK_GE(r, 0); // Fails only if fed invalid data. input_buffer_.clear(); - info->encoded_bytes = r; - info->encoded_timestamp = first_timestamp_in_buffer_; - info->payload_type = payload_type_; - // Allows Opus to send empty packets. - info->send_even_if_empty = true; - info->speech = r > 0; + EncodedInfo info; + info.encoded_bytes = r; + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = payload_type_; + info.send_even_if_empty = true; // Allows Opus to send empty packets. + info.speech = r > 0; + return info; } } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h index 0a2a008d1..bd76b4900 100644 --- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h +++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h @@ -58,11 +58,10 @@ class AudioEncoderOpus final : public AudioEncoder { bool dtx_enabled() const { return dtx_enabled_; } protected: - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; private: const int num_10ms_frames_per_packet_; diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc index 28c72fb3b..86f1158d9 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc @@ -60,48 +60,48 @@ void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) { speech_encoder_->SetProjectedPacketLossRate(fraction); } -void AudioEncoderCopyRed::EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) { - speech_encoder_->Encode(rtp_timestamp, audio, - static_cast(SampleRateHz() / 100), - max_encoded_bytes, encoded, info); +AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { + EncodedInfo info = speech_encoder_->Encode( + rtp_timestamp, audio, static_cast(SampleRateHz() / 100), + max_encoded_bytes, encoded); CHECK_GE(max_encoded_bytes, - info->encoded_bytes + secondary_info_.encoded_bytes); - CHECK(info->redundant.empty()) << "Cannot use nested redundant encoders."; + info.encoded_bytes + secondary_info_.encoded_bytes); + CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders."; - if (info->encoded_bytes > 0) { + if (info.encoded_bytes > 0) { // |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively // discarding the (empty) vector of redundant information. This is // intentional. - info->redundant.push_back(*info); - DCHECK_EQ(info->redundant.size(), 1u); + info.redundant.push_back(info); + DCHECK_EQ(info.redundant.size(), 1u); if (secondary_info_.encoded_bytes > 0) { - memcpy(&encoded[info->encoded_bytes], secondary_encoded_.get(), + memcpy(&encoded[info.encoded_bytes], secondary_encoded_.get(), secondary_info_.encoded_bytes); - info->redundant.push_back(secondary_info_); - DCHECK_EQ(info->redundant.size(), 2u); + info.redundant.push_back(secondary_info_); + DCHECK_EQ(info.redundant.size(), 2u); } // Save primary to secondary. - if (secondary_allocated_ < info->encoded_bytes) { - secondary_encoded_.reset(new uint8_t[info->encoded_bytes]); - secondary_allocated_ = info->encoded_bytes; + if (secondary_allocated_ < info.encoded_bytes) { + secondary_encoded_.reset(new uint8_t[info.encoded_bytes]); + secondary_allocated_ = info.encoded_bytes; } CHECK(secondary_encoded_); - memcpy(secondary_encoded_.get(), encoded, info->encoded_bytes); - secondary_info_ = *info; - DCHECK_EQ(info->speech, info->redundant[0].speech); + memcpy(secondary_encoded_.get(), encoded, info.encoded_bytes); + secondary_info_ = info; + DCHECK_EQ(info.speech, info.redundant[0].speech); } // Update main EncodedInfo. - info->payload_type = red_payload_type_; - info->encoded_bytes = 0; - for (std::vector::const_iterator it = - info->redundant.begin(); - it != info->redundant.end(); ++it) { - info->encoded_bytes += it->encoded_bytes; + info.payload_type = red_payload_type_; + info.encoded_bytes = 0; + for (std::vector::const_iterator it = info.redundant.begin(); + it != info.redundant.end(); ++it) { + info.encoded_bytes += it->encoded_bytes; } + return info; } } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h index 7ce9ca0f0..fd92d5245 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h @@ -45,11 +45,10 @@ class AudioEncoderCopyRed : public AudioEncoder { void SetProjectedPacketLossRate(double fraction) override; protected: - void EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - EncodedInfo* info) override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; private: AudioEncoder* speech_encoder_; diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index 2ae2fa21b..14c30d011 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc @@ -60,9 +60,8 @@ class AudioEncoderCopyRedTest : public ::testing::Test { void Encode() { ASSERT_TRUE(red_.get() != NULL); - encoded_info_ = AudioEncoder::EncodedInfo(); - red_->Encode(timestamp_, audio_, num_audio_samples_10ms, - encoded_.size(), &encoded_[0], &encoded_info_); + encoded_info_ = red_->Encode(timestamp_, audio_, num_audio_samples_10ms, + encoded_.size(), &encoded_[0]); timestamp_ += num_audio_samples_10ms; } @@ -83,18 +82,16 @@ class MockEncodeHelper { memset(&info_, 0, sizeof(info_)); } - void Encode(uint32_t timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded, - AudioEncoder::EncodedInfo* info) { + AudioEncoder::EncodedInfo Encode(uint32_t timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { if (write_payload_) { CHECK(encoded); CHECK_LE(info_.encoded_bytes, max_encoded_bytes); memcpy(encoded, payload_, info_.encoded_bytes); } - CHECK(info); - *info = info_; + return info_; } AudioEncoder::EncodedInfo info_; @@ -144,7 +141,8 @@ TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) { InSequence s; MockFunction check; for (int i = 1; i <= 6; ++i) { - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillRepeatedly(Return(AudioEncoder::kZeroEncodedBytes)); EXPECT_CALL(check, Call(i)); Encode(); check.Call(i); @@ -153,13 +151,13 @@ TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) { // Checks that no output is produced if the underlying codec doesn't emit any // new data, even if the RED codec is loaded with a secondary encoding. -TEST_F(AudioEncoderCopyRedTest, CheckNoOuput) { +TEST_F(AudioEncoderCopyRedTest, CheckNoOutput) { // Start with one Encode() call that will produce output. static const size_t kEncodedSize = 17; - AudioEncoder::EncodedInfo info; - info.encoded_bytes = kEncodedSize; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + AudioEncoder::EncodedInfo nonZeroEncodedBytes; + nonZeroEncodedBytes.encoded_bytes = kEncodedSize; + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(nonZeroEncodedBytes)); Encode(); // First call is a special case, since it does not include a secondary // payload. @@ -167,16 +165,14 @@ TEST_F(AudioEncoderCopyRedTest, CheckNoOuput) { EXPECT_EQ(kEncodedSize, encoded_info_.encoded_bytes); // Next call to the speech encoder will not produce any output. - info.encoded_bytes = 0; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(AudioEncoder::kZeroEncodedBytes)); Encode(); EXPECT_EQ(0u, encoded_info_.encoded_bytes); // Final call to the speech encoder will produce output. - info.encoded_bytes = kEncodedSize; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(nonZeroEncodedBytes)); Encode(); EXPECT_EQ(2 * kEncodedSize, encoded_info_.encoded_bytes); ASSERT_EQ(2u, encoded_info_.redundant.size()); @@ -192,8 +188,8 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes) { for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) { AudioEncoder::EncodedInfo info; info.encoded_bytes = encode_size; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) - .WillOnce(SetArgPointee<4>(info)); + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) + .WillOnce(Return(info)); } // First call is a special case, since it does not include a secondary @@ -218,7 +214,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) { helper.info_.encoded_bytes = 17; helper.info_.encoded_timestamp = timestamp_; uint32_t primary_timestamp = timestamp_; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) .WillRepeatedly(Invoke(&helper, &MockEncodeHelper::Encode)); // First call is a special case, since it does not include a secondary @@ -249,7 +245,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloads) { payload[i] = i; } helper.payload_ = payload; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) .WillRepeatedly(Invoke(&helper, &MockEncodeHelper::Encode)); // First call is a special case, since it does not include a secondary @@ -286,7 +282,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) { helper.info_.encoded_bytes = 17; const int primary_payload_type = red_payload_type_ + 1; helper.info_.payload_type = primary_payload_type; - EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _, _)) + EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)) .WillRepeatedly(Invoke(&helper, &MockEncodeHelper::Encode)); // First call is a special case, since it does not include a secondary diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc index 3a6a6efa2..a47dbdab0 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc @@ -234,8 +234,8 @@ void ACMGenericCodec::Encode(uint32_t input_timestamp, first_frame_ = false; CHECK_EQ(audio_channel, encoder_->NumChannels()); - encoder_->Encode(rtp_timestamp_, audio, length_per_channel, - 2 * MAX_PAYLOAD_SIZE_BYTE, bitstream, encoded_info); + *encoded_info = encoder_->Encode(rtp_timestamp_, audio, length_per_channel, + 2 * MAX_PAYLOAD_SIZE_BYTE, bitstream); *bitstream_len_byte = static_cast(encoded_info->encoded_bytes); } diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc index e319b000f..9f3b0fed1 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -150,9 +150,9 @@ class AudioDecoderTest : public ::testing::Test { samples_per_10ms, channels_, interleaved_input.get()); - audio_encoder_->Encode(0, interleaved_input.get(), - audio_encoder_->SampleRateHz() / 100, - data_length_ * 2, output, &encoded_info_); + encoded_info_ = audio_encoder_->Encode( + 0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100, + data_length_ * 2, output); } EXPECT_EQ(payload_type_, encoded_info_.payload_type); return static_cast(encoded_info_.encoded_bytes);